Has anyone here got Sipgate working with Trixbox?

I am banging my head against a wall here!

If I call my Sipgate number form my normal phone, I am getting the message
(From my trixbox) "The number you have dialled is not in service, please
check the number and try again" beep beep beep etc...

From a fresh install I have done the following...

I was previously on the last build of Asterisk@home without problem

---------

In sip.conf, added the following...
externip=MY IP ADDRESS
outside_addr=MY IP ADDRESS

---------

Added my extensions, 200 and 201 - these can call each other.

---------

Added a new trunk with the following details (Copied from my working
Asterisk@home setup)

Outbound Caller ID - MY SIPGATE NUMBER
Max Channels - 1
Dial Rules - 0044+XXXXXXXXXX (Plus a load more other rules)
Trunk Name - Sipgate
PEER Details - allow=ulaw&alaw
authuser=SIPGATE ID
canreinvite=no
context=ext-did
disallow=all
dtmfmode=info
fromdomain=sipgate.co.uk
fromuser=SIPGATE ID
host=sipgate.co.uk
insecure=very
qualify=yes
secret=SIPGATE PASSWORD
type=peer
username=SIPGATE ID

Register String - SIPGATE ID:PASSWORD@sipgate.co.uk/SIPGATE ID

--------------------

Inbound Routes

DID Number - SIPGATE ID

Destination - Office <200>

-------------------


Any ideas what else I need to do!

Here is what the terminal spews out when I call my sipgate PSTN number from
a normal phone

(I have change my sipgate ID to MY SIPGATE ID)


-- Executing NoOp("SIP/gw02.uk.sipgate.net-09b0c6f8", "Received incoming
SIP connection from unknown peer to MY SIPGATE ID") in new stack
-- Executing Set("SIP/gw02.uk.sipgate.net-09b0c6f8", "DID=MY SIPGATE
ID") in new stack
-- Executing Goto("SIP/gw02.uk.sipgate.net-09b0c6f8", "s|1") in new
stack
-- Goto (from-sip-external,s,1)
-- Executing GotoIf("SIP/gw02.uk.sipgate.net-09b0c6f8", "0?from-trunk|MY
SIPGATE ID|1") in new stack
-- Executing Set("SIP/gw02.uk.sipgate.net-09b0c6f8",
"TIMEOUT(absolute)=15") in new stack
-- Channel will hangup at 2006-08-04 14:31:06 UTC.
-- Executing Answer("SIP/gw02.uk.sipgate.net-09b0c6f8", "") in new stack
-- Executing Wait("SIP/gw02.uk.sipgate.net-09b0c6f8", "2") in new stack
-- Executing Playback("SIP/gw02.uk.sipgate.net-09b0c6f8",
"ss-noservice") in new stack
-- Playing 'ss-noservice' (language 'en')
-- Executing Congestion("SIP/gw02.uk.sipgate.net-09b0c6f8", "") in new
stack

So the call is getting to asterisk, but it is refusing to route it!

Thanks!!

Sparks...